did you notice the two dots in the IP address of ldaphost ?
Marco.
Chandan Mishra wrote:
Hi
I want to authenticate the asterisk users from the LDAP directory server
not from the sip.conf.
I tried to use the astirectory-1.2
http://www..asterisk-ev.org/download/astirectory-1.2-0.3.tgz . But
Hi,
I enabled Callprogress in the zapata.conf , so in the CDR it will log
other things other then answered (Busy, no answer etc),
but, this seems to break my Polycom's DTMF, i configured RFC2833 for the
dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt
reach the other
Hi
I enabled Callprogress in the zapata.conf , so in the CDR it will log
other things other then answered (Busy, no answer etc),
but, this seems to break my Polycom's DTMF, i configured RFC2833 for the
dtmf in the sip.conf, and when callprogress is enabled, the dtmf doesnt
reach the other end,
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dial's in, when hanging up, the asterisk will
detect the hangup only after 10 seconds, i searched around, and found
many similar problems, but no solution, i tried some options in
Hi,
I have a problem with the Caller ID string, seems like asterisk will
display only 10 digits of the caller id.
If the string is longer then 10 digits, asterisk will sometimes strip
the first digit, and sometimes the last digits, in order to show a
10-digit callerid,
Is this
Yes, didnt change anything
Marco.
Angelito Manansala wrote:
hmmm
di you try this ;hanguponpolarityswitch=yes
Cheerz!
On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote:
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dial's
Hi,
I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1
cards, is this possible ? and do i need any other modules except for the
E1 modules ?
What i want to do is connect the asterisk to the PRI through the Cisco
router, and let my legacy PBX utilize some of the PRI
Hi,
Is there a way to detect (in the dialplan) if a SIP peer is registered
with the server ?
I am using macros to dial to extension, becuase i dont want to define
each extension in the dialplan, and, for example, my numbers are 8xx , i
want to know if a peer exists/registered before
Hi,
I am using Asterisk 1.0.9 with the 1.2.0 zaptel, just for the fxotune
utility, which solved my echo problems , my zttest results are low, but
no echo on ZAP lines...
Marco.
Chris Miller wrote:
Mojo with Horan Company, LLC wrote:
The recent suggestion on the list was to not use
Hi,
As anyone tried integrating App_Directory with any Text2Speech mechanism
like festival ?
Marco.
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Yes, i am having timeouts on registering to the LAX sip server of
broadvoice.
Marco.
Nate Kapi wrote:
I've been having a lot of problems with Broadvoice lately. Anyone else
been without service for extended periods of time this week?
___
Hi,
I would like to know what type of configuration could get me closer to
100% hits in zttest, when testing a TDM400P with 4 FXO ports,
I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh
CPU, HT is disabled, PCI latency was changed, i still cant get more then
99.975% in
2911
f +64 9 358 9070
m +64 21 415 297
w www.fff.co.nz
Marco Supino wrote:
Hi,
I would like to know what type of configuration could get me closer to
100% hits in zttest, when testing a TDM400P with 4 FXO ports,
I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh
CPU, HT
Hi,
I asked yesterday about a problem with x306 and IRQ sharing, didnt get
much info, now, i was playing with lspci, and see something strange,
lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is
also on IRQ 7,
lspci -bv (from the man - b - shows bus-centric view, as seen by
Hi,
This is a little off-topic,but if someone has any info, it could help me
a LOT!,
I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my
problem is that the BIOS assigns the same IRQ to the SCSI controller,
and the TDM400P, i have tried several options of making the bios
Of
Marco Supino
Sent: Saturday, September 24, 2005 8:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] IBM x306
Hi,
This is a little off-topic,but if someone has any info, it
could help me a LOT!,
I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI
machine,my problem
Hi,
I tried setpci INTERRUPT_LEVEL (or something similar, cant remmeber
now), and also setpci seems like it changed the IRQ, lspci -v still
shows the old IRQ
Marco.
Stefan de Konink wrote:
On Sun, 25 Sep 2005, Marco Supino wrote:
I am building an asterisk pbx (1.0.9) on an IBM x306
Hi,
Anyone has a working example of VoiceXML with asterisk ? i was looking
around voip-info and the internet, and couldnt find more then proof of
concept documents.
Also, does anyone knows how FWD does their VoiceXML (411) service ?
Thanks for any info
Marco.
Hi,
I was looking for solutions for simple FXO cards, and came across the
two modem channels in the asterisk channels/ dir, i assume they are
there becuase someone made these two types of modems work as FXO (or are
they there for other purpose ?),
does anyone have any info on these channels ?
Hi,
I need some info from people with the x100p card (digium or clone),
please send me the output of lspci and lspci -n from your linux
machine, i am tring to find out something on my * server.
Thanks.
Marco.
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Do you have an 's' extention in the default context ?
Marco.
Dimitris Kounalakis wrote:
Hello,
I am trying to configure asterisk 1.0.7pre to get incoming calls from an
ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
that the context is not recognised in the
Hi,
I am adding phones to my asterisk setup, until now i worked with some
softphones, with no problem,
I got some Grandstream BT100 phones, and see something strange in the
log, the on the phone's screen,
This is from the log :
Found peer '122'
Looking for 122 in default
Transmitting (no NAT):
Hi,
I tried to add the IAXTel config to my asterisk, so i can dial free
numbers inside the US from my SIP softphone (X-lite), everything seems
to be working, but the sound quality is terrible, the other side sounds
like a digitized voice, and the voice is cut, i cant hear a full word,
I tried
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